signal-processing

Fast Fourier Transform for non log 2

亡梦爱人 提交于 2020-01-01 20:00:08
问题 I want to implement Fast Fourier Transform in Java for chord recognition, but I don't really get it. It says that the number of samples should be a power of 2, so what should we do for a song that doesn't have number of samples equal to a power of 2? Also I would like to know about the STFT. 回答1: You normally generate an STFT over a sliding window throughout your file. The size of the window is chosen to give a reasonable time period over which the characteristics of the sound do not change

Objective-C Peak Detection Accelerate Framework

十年热恋 提交于 2020-01-01 19:57:34
问题 I am a no math guru here, so I want to ask anyone familiar with Digital Signal Processing, what is the best way of detecting real time peaks. I get about 30 frames/values a second and I've tried to implement the slope algorithm for detecting peaks, it worked OK, about 80% of the cases, but its really not good enough :(. From what I've searched one should use the Fast Fourier Transform, but I have no idea how to get started with it, perhaps I'm missing the general idea of how I should use FFT

Goertzel algorithm to get the phase?

天涯浪子 提交于 2020-01-01 17:11:45
问题 I am using Goertzel algorithm to get the amplitude of a certain frequency. I am trying now to get the phase from it, and I don't know how. Can some one explain, and show me how to get the phase of a certain-f from this code? Also, I am using it to 16khz, with sample rate 44.1. What's the smallest length of samples that I can run it on? double AlgorithmGoertzel( int16_t *sample,int sampleRate, double Freq, int len ) { double realW = 2.0 * cos(2.0 * M_PI * Freq / sampleRate); double imagW = 2.0

Voice Alteration Algorithm

随声附和 提交于 2020-01-01 07:02:54
问题 Could somebody point me to a voice alteration algorithm? Preferably in Java or C? Something that I could use to change a stream of recorded vocals into something that sounds like Optimus Prime. (FYI- Optimus Prime is the lead Autobot from transformers with a very distinctive sounding voice... not everybody may know this.) Is there an open source solution? 回答1: You can't just change the sample rate. The human voice has formants. What you want to do is move the formants. That should be your

Result from processing audio signal with Goertzel algorithm

柔情痞子 提交于 2020-01-01 06:01:47
问题 I made a little signal processing app. It processes audio signal (morse code) on certain frequency with Goerztel algorithm. Application saves temporary file to the filesystem and after recording is finished, starts to detect signals. Now I got the result with bunch of magnitudes. I don't really know what to read from those magnitudes. How can I decode the morse code from those magnitudes? How can I read them? Tried to find references, but nowhere is explained what is the result and how to

Result from processing audio signal with Goertzel algorithm

こ雲淡風輕ζ 提交于 2020-01-01 06:00:08
问题 I made a little signal processing app. It processes audio signal (morse code) on certain frequency with Goerztel algorithm. Application saves temporary file to the filesystem and after recording is finished, starts to detect signals. Now I got the result with bunch of magnitudes. I don't really know what to read from those magnitudes. How can I decode the morse code from those magnitudes? How can I read them? Tried to find references, but nowhere is explained what is the result and how to

Frequency response using FFT in MATLAB

限于喜欢 提交于 2020-01-01 03:38:24
问题 Here is the scenario: using a spectrum analyzer i have the input values and the output values. the number of samples is 32000 and the sampling rate is 2000 samples/sec, and the input is a sine wave of 50 hz , the input is current and the output is pressure in psi. How do i calculate the frequency response from this data using MATLAB, using the FFT function in MATLAB. i was able to generate a sine wave, that gives out the the magnitude and phase angles, here is the code that i used: %FFT

Swift FFT - Complex split issue

安稳与你 提交于 2019-12-31 13:15:07
问题 I am trying to perform FFT on an audio file to find frequency using the Accelerate framework. I have adapted code (probably wrong) from this question: Spectrogram from AVAudioPCMBuffer using Accelerate framework in Swift Although, the magnitudes from ' spectrum ' are either ' 0 ', ' inf ' or ' nan ', and the ' real ' and ' imag ' components of the complex split print similar results; indicating that this is the cause of the problem as: ' magnitude = sqrt(pow( real ,2)+pow( imag ,2) '. Correct

calculate turning points / pivot points in trajectory (path)

痞子三分冷 提交于 2019-12-31 08:18:21
问题 I'm trying to come up with an algorithm that will determine turning points in a trajectory of x/y coordinates. The following figures illustrates what I mean: green indicates the starting point and red the final point of the trajectory (the entire trajectory consists of ~ 1500 points): In the following figure, I added by hand the possible (global) turning points that an algorithm could return: Obviously, the true turning point is always debatable and will depend on the angle that one specifies

calculate turning points / pivot points in trajectory (path)

拟墨画扇 提交于 2019-12-31 08:15:15
问题 I'm trying to come up with an algorithm that will determine turning points in a trajectory of x/y coordinates. The following figures illustrates what I mean: green indicates the starting point and red the final point of the trajectory (the entire trajectory consists of ~ 1500 points): In the following figure, I added by hand the possible (global) turning points that an algorithm could return: Obviously, the true turning point is always debatable and will depend on the angle that one specifies